1. Field of the Invention
The present invention relates to voice signal encoding and decoding apparatus and methods suitable for use in various communication devices,. etc., and more particularly to voice signal encoding and decoding apparatus and methods which rapidly attenuate an erroneous influence when there is an error in transmission and provides robustness to such error in transmission.
2. Description of the Related Art
FIG. 1 shows a proposed voice signal encoding and decoding apparatus which is called an voice codec and which includes an encoding section 1 and a decoding section 2. In FIG. 1, the encoding section 1 of a transmitter side and the decoder section 2 of a receiver side only are shown, but actually, the transmitter also has a decoding section similar to that of the receiver and the receiver also has an encoding section similar to that of the transmitter. The encoding section 1 of the transmitter and the decoding section 2 of the receiver are connected via a transmission channel 3. The encoding section 1 includes an input terminal 4, a buffer 5, a predicted parameter computer 6, a subtractor 7, a quantizer 8, an inverse-quantizer 33, an adder 9, a predictive filter 10, encoders 11, 13 and a multiplexer (MPX) 12.
An input voice signal x received at the input terminal 4 is delivered to the buffer 5 where only a predetermined number of processed samples of the input signal is stored. The predicted parameter computer 6 computes a predicted parameter which eliminates the stochastic redundancy of the input voice signal x by using the predetermined number of samples of the input voice signal x stored in the buffer 5. The output signal of the buffer 5 is applied to the subtractor 7 which subtracts from the output signal of the buffer 5 a predicted value x output of the predictive filter 10 to provide a prediction difference d. The output signal of the subtractor 7 is applied to the quantizer 8 which quantizes the prediction difference d from the subtractor 7. The output d of the quantizer 8 is applied via the inverse-quantizer 33 to the adder 9, which adds the quantized prediction difference d' and the predicted value x to produce a locally decoded signal x. The output signal x is applied to the predictive filter 10.
The predictive filter 10 receives the predictive parameter calculated by the predicted parameter computer 6 and computes a predicted value x in accordance with the signal x and the predicted parameter.
The output d from the quantizer 8 is applied to the encoder 11, which encodes the quantized difference d.
The output from the predicted parameter computer 6 is applied to the encoder 13, which encodes a predicted parameter output by the predicted parameter computer 6. The multiplexer 12 produces formatted data, as shown in FIG. 2, from the data encoded by the encoders 11 and 13. As shown in FIG. 2, the data includes a synchronizing pattern 19, a side information section 20 and a main information section 21. The synchronizing pattern 19 shows the beginning of a frame. The side information section 20 includes data such as the predicted parameter, power and the fundamental voice frequency (pitch), etc. The main information section 21 includes the prediction difference.
The decoding section 2 includes a demultiplexer (DEMPX) 14, decoders 15, 15a, and an adder 16, a predictive synthesizing filter 17, and an output terminal 18. The demultiplexer 14 divides into sub-fields a frame received through the transmission channel 3. The decoder 15 decodes data existing in a divided sub-field and outputs a prediction difference d'. The decoder 15a decodes a predicted parameter.
The adder 16 adds the prediction difference d' and a predicted value x produced by the predictive synthesizing filter 17 to produce an output voice signal x', which is then output from the output terminal 18.
The predictive synthesizing filter 17 generates a predicted value x' from the output voice signal x' and the predicted parameter decoded by the decoder 15a.
In summary, the predicted value x is subtracted by the subtractor 7 from the input voice signal stored in the buffer 5 and the resulting prediction difference d is input to and quantized by the quantizer 8. The output from the quantizer is encoded by the encoder 11, and output by the multiplexer 12 as frame data such as shown in FIG. 2 to the transmission channel 3.
The frame data received by the transmission channel 3 is divided by the demultiplexer 14 into respective sub-fields data segments, which are then decoded by the decoder 15. The predicted value x' is added by the adder 16 to the output from the decoder 15, and the resulting signal is output as an output voice signal x' to the output terminal 18.
If an error occurs in transmission in the use of such conventional voice codec, noise or rasping voice is output from the decoder 2. The influence of such error will appear on the outputs in the future as well as at present. Therefore, an attempt has been made to produce the silence when there occurs a transmission error, but a sensation of hearing is not good.
Namely, the conventional voice codec is vulnerable to a transmission error and the influence of the error tends to continue still in the future.
The present invention perceives such problems. It is an object of the present invention to provide an voice signal encoding and decoding apparatus and method which has great robustness to errors in transmission.